--- title: "Troubleshooting Voice Quality and Network Configuration" slug: "troubleshooting-voice-quality-and-network-configuration" updated: 2026-07-11T00:01:06Z published: 2026-07-11T00:01:06Z canonical: "help.servicetitan.com/troubleshooting-voice-quality-and-network-configuration" --- > ## Documentation Index > Fetch the complete documentation index at: https://help.servicetitan.com/llms.txt > Use this file to discover all available pages before exploring further. # Troubleshooting Voice Quality and Network Configuration ## Problem If voice calls on this connection sound rough—choppy audio, missed words, pauses, or dropped calls—your network is likely the cause. These symptoms can also appear as a Fair or Poor result when you run a network test. Networks vary—your office network and your home network are different, and even the same network changes through the day. A poor result usually points to something on your local network, your computer, or the path between you and your internet provider, rather than ServiceTitan. Work through the steps below in order, and re-run a network test after each one. If the result improves, no further troubleshooting is needed. ## Solution ### Agent troubleshooting Start with the first three workflows. They address the most common causes of voice quality issues. #### 1. Switch from Wi-Fi to a wired (Ethernet) connection A wired connection is often more stable than Wi-Fi for voice calls. Wi-Fi can add inconsistency that affects voice quality, even when speed tests look fine. 1. Plug a **Cat5e** or **Cat6** Ethernet cable from your computer directly into a wall jack, switch, or router. 2. If your laptop does not have an Ethernet port, use a USB-C or USB-A Ethernet adapter. 3. Re-run the network test. If a wired connection is not available, move as close to your router as possible. Use the **5 GHz Wi-Fi band** instead of **2.4 GHz**. #### 2. Reduce competing traffic Competing traffic can reduce the bandwidth available for voice calls. This can happen when uploads, streams, backups, or downloads are running on the same network. 1. Close browser tabs and apps that use a lot of bandwidth, such as video, Zoom, Google Meet, file backups, Dropbox or OneDrive sync, and large downloads. 2. Ask other people on the network to pause streaming or uploading for a few minutes. 3. Re-run the network test. #### 3. Restart your modem and router Restarting your modem and router can clear temporary connection issues. 1. Unplug the modem and router for 30 seconds. 2. Plug the modem back in first. 3. Wait until the modem lights settle. This usually takes 1–2 minutes. 4. Plug the router back in. 5. Re-run the network test. #### 4. Disconnect from your VPN Contact Center Pro is not supported over a Virtual Private Network (VPN). 1. Disconnect from the VPN. 2. Re-run the network test. If the result improves after disconnecting from the VPN, ask your IT team to configure Split tunneling. Split tunneling lets ServiceTitan voice traffic bypass the VPN while other traffic can continue through it. See the section for IT below. #### 5. Check your cables Damaged or outdated cables can affect voice quality. 1. Check that Ethernet cables say **Cat5e** or **Cat6** on the side. 2. Replace cables labeled **Cat5**. 3. Look for chewed, kinked, pinched, or damaged cables. 4. Check for broken plastic clips. 5. Plug the cable into a different port on the switch or router, if one is available. 6. Re-run the network test. #### 6. Check your computer and headset Computer performance and audio device settings can affect call quality. 1. Confirm that your headset is charged. 2. Confirm that your headset is selected as the input and output device in your browser. 3. Check your *CPU* usage: 1. On Windows: 1. Press **Ctrl + Shift + Esc** on your keyboard. 2. Click **Task Manager > More details > Performance** tab. 3. Go to the **CPU** tab. 2. On MAC: 1. Press **Command + Spacebar** to open *Spotlight*, type Activity Monitor, and press Enter. 2. Click the CPU tab at the top of the window. 4. If CPU usage is near 100%, close unused apps or restart the computer before testing again. 5. Close unused browser tabs and background apps. 6. Restart the computer if it has been running for several days. 7. Make sure your operating system and audio drivers are up to date. 8. Re-run the network test. #### 7. Use a supported internet connection ServiceTitan does not support satellite (Starlink, HughesNet, Viasat) or mobile/cellular connections (hotspots, phone tethering, 4G/5G modems) for Contact Center Pro. Satellite latency is too high for real-time voice. Cellular connections degrade unpredictably as tower congestion and signal strength change. Switch to a standard home or office internet connection. Broadband from a cable, phone, or fiber provider all work—if you're not sure which you have, check that your modem plugs into a wall jack rather than picking up a wireless signal. --- ### IT team or network admin troubleshooting If the steps above didn't resolve the issue, the remaining causes typically require your IT team or network administrator. ServiceTitan does not configure customer networks—network configuration changes must be made by your IT team, network administrator, or Internet Service Provider (ISP). #### Disable SIP ALG Session Initiation Protocol Application Layer Gateway (SIP ALG) is a router feature that modifies SIP packets in transit, ostensibly for Network Address Translation (NAT) traversal. In practice, it corrupts SIP headers, rewrites Contact/Via fields, and interferes with Secure Real-time Transport Protocol (SRTP) key exchange. Symptoms include call setup failures, one-way audio, calls dropping after about 30 seconds, and registration failures. Disable it in router/firewall settings: - **SonicWall:** Manage > VoIP → Settings → uncheck **Enable SIP Transformations** - **Cisco / Meraki:** Security Appliance → Firewall → Layer 7 → disable **SIP inspection** - **Fortinet:** System → Feature Visibility → disable **SIP ALG helper** - **Palo Alto / generic NGFW:** Disable the SIP application layer gateway in the security profile After disabling, reboot the router and have the agent re-run the test. > [!NOTE] > **Note:** Not all gateways expose a SIP ALG setting. On some (for example, T-Mobile remote gateways), the feature cannot be disabled by the customer. On others, the setting is locked from end-user access. In those cases, contact your ISP and ask them to disable SIP ALG. For more, see [Disable SIP ALG to improve ServiceTitan call quality](/v1/docs/disable-sip-alg-to-improve-servicetitan-call-quality). #### Configure split tunneling for VPN Contact Center Pro is not supported over a full-tunnel VPN. Configure split tunneling so traffic to ServiceTitan voice servers bypasses the VPN. This is the most common cause of high-latency results in offices that otherwise look healthy. #### Verify ISP performance from the agent's machine Run a speed test directly from the agent's computer through [fast.com](http://fast.com) or [speedtest.net](http://speedtest.net). Voice doesn't require much bandwidth. It requires about 100 kbps in each direction per call. However, that bandwidth must be reliably available and not temporarily consumed by other office traffic. For an office of *N* concurrent agents, plan for at least: - **Upload:** *N* × 100 kbps + 50% headroom - **Download:** The same, plus normal office traffic If the measured speed is far below the ISP plan rate, the issue is ISP-side. Escalate to the provider, not to ServiceTitan. #### Configure QoS / DSCP marking For LANs with mixed voice and bulk traffic: - Mark RTP traffic with **DSCP EF (46)** at the LAN edge. - Verify switches honor DSCP and do not strip it. - Enforce QoS policies at the WAN edge router. - For Wi-Fi, ensure **WMM** (Wi-Fi Multimedia) is enabled and prefer the 5 GHz band. #### Verify firewall ports Two sets of ports matter, depending on whether you're running a network test or carrying live voice traffic. UDP 5060 appears in both—it's required for the test and for live voice. For the network test (Visualware): - UDP 5060 - UDP 8090 - TCP 20000, 20001 - UDP 20000, 20001 For live Contact Center Pro voice traffic: | **Port** | **Protocol** | **Function** | | --- | --- | --- | | 5060 | UDP/TCP | SIP signaling | | 5061 | TCP | SIP TLS (secure signaling) | | 7000 | UDP/TCP | Kazoo WebSocket signaling | | 16384–32768 | UDP | RTP media (audio stream) | | 80, 443, 8443 | TCP | HTTP/HTTPS, admin/API | Outbound to all listed ports must be permitted. If you are running a next-generation firewall with application-ID inspection, ensure SIP and Real-Time Transport Protocol (RTP) application IDs are in the allow policy—port-based rules alone may not be sufficient if app-ID inspection is silently dropping traffic. --- ## When to contact ServiceTitan support Reach out if any of these are true: - You worked through the steps above (including the IT-team section, if applicable) and your result is still *Poor*. - All your agents at the same office are getting *Fair* or *Poor* at the same time, and Wi-Fi/VPN are already ruled out. - The test result is *Good* but agents are still hearing problems on real calls. - Calls connect but one side hears silence, even after disabling SIP ALG. When you open a ticket, include: - A screenshot of the network test results screen, including the Reference ID. - The affected agent’s name. - The approximate date, time, and time zone when the issue happened. - Whether the agent is on Wi-Fi or a wired connection. - Whether the agent is connected through a VPN. - Whether split tunneling is configured, if the agent uses a VPN. Also include the following details if you have them: - An example phone number from a recent affected call. - Your firewall or router make and model. - A current speed test result. - Any recent network changes, such as a new ISP, new router, or office move. ## Reference ### What "Fair" and "Poor" mean - **Good**: Calls sound clear and natural. No action needed. - **Fair**: Calls mostly work, but you may notice short audio glitches, brief delays, or a word dropping out here and there. Most short calls will still feel fine; long or back-to-back calls are where you'll feel it. - **Poor**: Calls have noticeable problems—choppy or robotic audio, missing words, awkward pauses, talking over each other, or calls dropping. Calls on this connection will be hard for both the agent and the customer. ### What the test measured The test scored four things and showed you the worst result of the four. Packet Loss, Jitter, and Latency are the three direct measurements. MOS is an overall grade calculated from those three—if your Mean Opinion Score (MOS) is low, look at which of the other three is driving it down. | **Metric** | **Good** | **Fair** | **Poor** | | --- | --- | --- | --- | | Packet Loss | <1% | 1–3% | >3% | | Jitter | <30 ms | 30–100 ms | >100 ms | | Latency (RTT) | <150 ms | 150–300 ms | >300 ms | | MOS | ≥3.6 | 3.0–3.5 | <3.0 | - **Packet Loss**: How many pieces of your audio went missing on the way. Lost audio cannot be re-sent on a live call—it just disappears. - **Jitter**: How consistent the connection is. Inconsistent connections cause choppy or robotic audio even when speeds look fast. - **Latency**: The delay between you and the server. High latency causes that "satellite call" feeling where you keep talking over each other. - **MOS**: An overall 1–5 quality score (like a report-card grade) calculated from the three above. ### Identify symptoms and patterns The network test identifies symptoms, such as high latency, but it may not identify the underlying cause. Issues can come from a faulty cable, congested Wi-Fi channel, nearby interference, or another network condition. You can re-run the network test as many times as needed. A single Fair or Poor result may be a temporary spike. Re-run the test during the time of day when calls sound worst. If the result is still Fair or Poor, the issue is persistent and worth investigating. If the result is only Fair or Poor sometimes, include that information when you contact ServiceTitan Support. ## Want to learn more? - See [Disable SIP ALG to improve ServiceTitan call quality](/v1/docs/disable-sip-alg-to-improve-servicetitan-call-quality) - See [Best practices for optimal VoIP experience and reliability](/v1/docs/best-practices-for-optimal-voip-experience-and-reliability) - See [Contact Center Pro network recommendations and setup](/v1/docs/contact-center-pro-network-recommendations-and-setup) ServiceTitan's enterprise-grade call center product, adding advanced queue management, live coaching, and AI features to Phones Pro. **What it does:** Enables multi-agent contact center operations with real-time supervision, analytics, and AI-assisted call handling at scale. **Also called:** CCP, Advanced Phones Pro **Tags:** Contact Center Pro [See the full glossary →](https://help.servicetitan.com/docs/glossary-overview) **a.k.a.** CCP Split determines how much credit each technician is getting for the job. For example: If two technicians perform work on a job, the split could be an even 50%-50%, or it could be 75%-25% if the first technician did a bulk of the work. By default 100% of the technician split will go to the first technician assigned to the job. [Learn more →](https://help.servicetitan.com/docs/servicetitan-glossary) · [See the full glossary →](https://help.servicetitan.com/docs/glossary-overview)